webrtcbin: fix pt selection for FEC and RTX when BUNDLE

When we offer bundled media, payload types must be unique
across all bundled media, as they will be multiplexed in the
same session.

1 files changed, 47 insertions, 5 deletions

diff --git a/ext/webrtc/gstwebrtcbin.c b/ext/webrtc/gstwebrtcbin.c
index 76c99edf0..9236753d1 100644
--- a/ext/webrtc/gstwebrtcbin.c
+++ b/ext/webrtc/gstwebrtcbin.c
@@ -1946,7 +1946,8 @@ _add_fingerprint_to_media (GstWebRTCDTLSTransport * transport,
 static gboolean
 sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
     GstWebRTCRTPTransceiver * trans, GstWebRTCSDPType type, guint media_idx,
-    GString * bundled_mids, guint bundle_idx, gboolean bundle_only)
+    GString * bundled_mids, guint bundle_idx, gboolean bundle_only,
+    GArray * reserved_pts)
 {
   /* TODO:
    * rtp header extensions
@@ -2028,14 +2029,14 @@ sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
   }
 
   if (type == GST_WEBRTC_SDP_TYPE_OFFER) {
-    GArray *reserved_pts = g_array_new (FALSE, FALSE, sizeof (guint));
     const GstStructure *s = gst_caps_get_structure (caps, 0);
     gint clockrate = -1;
     gint rtx_target_pt;
     gint original_rtx_target_pt;        /* Workaround chrome bug: https://bugs.chromium.org/p/webrtc/issues/detail?id=6196 */
     guint rtx_target_ssrc = -1;
 
-    if (gst_structure_get_int (s, "payload", &rtx_target_pt))
+    if (gst_structure_get_int (s, "payload", &rtx_target_pt) &&
+        webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE)
       g_array_append_val (reserved_pts, rtx_target_pt);
 
     original_rtx_target_pt = rtx_target_pt;
@@ -2054,7 +2055,6 @@ sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
     if (original_rtx_target_pt != rtx_target_pt)
       _pick_rtx_payload_types (webrtc, WEBRTC_TRANSCEIVER (trans), reserved_pts,
           clockrate, original_rtx_target_pt, rtx_target_ssrc, media);
-    g_array_free (reserved_pts, TRUE);
   }
 
   _media_add_ssrcs (media, caps, webrtc, WEBRTC_TRANSCEIVER (trans));
@@ -2084,6 +2084,34 @@ sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
   return TRUE;
 }
 
+static void
+gather_pad_pt (GstWebRTCBinPad * pad, GArray * reserved_pts)
+{
+  if (pad->received_caps) {
+    GstStructure *s = gst_caps_get_structure (pad->received_caps, 0);
+    gint pt;
+
+    if (gst_structure_get_int (s, "payload", &pt)) {
+      g_array_append_val (reserved_pts, pt);
+    }
+  }
+}
+
+static GArray *
+gather_reserved_pts (GstWebRTCBin * webrtc)
+{
+  GstElement *element = GST_ELEMENT (webrtc);
+  GArray *reserved_pts = g_array_new (FALSE, FALSE, sizeof (guint));
+
+  GST_OBJECT_LOCK (webrtc);
+  g_list_foreach (element->sinkpads, (GFunc) gather_pad_pt, reserved_pts);
+  g_list_foreach (webrtc->priv->pending_pads, (GFunc) gather_pad_pt,
+      reserved_pts);
+  GST_OBJECT_UNLOCK (webrtc);
+
+  return reserved_pts;
+}
+
 /* TODO: use the options argument */
 static GstSDPMessage *
 _create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options)
@@ -2093,6 +2121,7 @@ _create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options)
   GString *bundled_mids = NULL;
   gchar *bundle_ufrag = NULL;
   gchar *bundle_pwd = NULL;
+  GArray *reserved_pts = NULL;
 
   gst_sdp_message_new (&ret);
 
@@ -2115,6 +2144,7 @@ _create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options)
 
   if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE) {
     _generate_ice_credentials (&bundle_ufrag, &bundle_pwd);
+    reserved_pts = gather_reserved_pts (webrtc);
   }
 
   /* for each rtp transceiver */
@@ -2136,6 +2166,7 @@ _create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options)
 
     /* FIXME: only needed when restarting ICE */
     if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) {
+      reserved_pts = g_array_new (FALSE, FALSE, sizeof (guint));
       _generate_ice_credentials (&ufrag, &pwd);
     } else {
       ufrag = g_strdup (bundle_ufrag);
@@ -2147,8 +2178,11 @@ _create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options)
     g_free (ufrag);
     g_free (pwd);
 
+    g_assert (reserved_pts != NULL);
+
     if (sdp_media_from_transceiver (webrtc, &media, trans,
-            GST_WEBRTC_SDP_TYPE_OFFER, i, bundled_mids, 0, bundle_only)) {
+            GST_WEBRTC_SDP_TYPE_OFFER, i, bundled_mids, 0, bundle_only,
+            reserved_pts)) {
       if (bundled_mids) {
         const gchar *mid = gst_sdp_media_get_attribute_val (&media, "mid");
 
@@ -2159,6 +2193,14 @@ _create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options)
     } else {
       gst_sdp_media_uninit (&media);
     }
+
+    if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) {
+      g_array_free (reserved_pts, TRUE);
+    }
+  }
+
+  if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE) {
+    g_array_free (reserved_pts, TRUE);
   }
 
   /* add data channel support */